When the audioconverter needs more data, let it return NEED_DATA. This
can happen before the ports actually have consumed all the input data.
For example, then the next cycle would require 1024 samples but there
are currently only 16 samples queued, the next cycle will consume the
16 samples and then need another buffer to produce output.
For rt streams, this is not a problem because a new buffer will be
fetched in the next cycle synchronously.
When the stream is async, we can use this NEED_DATA to prefetch a
new buffer so that we have one in the next cycle.
This fixes hickups with async streams that provide random sized
buffers.
Move the setup of the output buffers first.
Then figure out how many samples we need to produce and consume.
Make sure we use the resampler to only convert the input samples that
are needed to produce the output samples.
Fixes some muddled sound with mpv when upmixing.
Remove the redundant remap array.
First set up the array with remapped output pointers in case we
can do passthrough output. Make sure stages write to the remapped
array in passthrough.
Remap the input array after unpack/convert.
This avoid an input and output memcpy in the common case of
remapping.
When we are actively driving the stream and the converter needs more
data, call the stream process function again to get it so that we
don't underrun.
Fixes#2494
Tweak the conversion constants a bit so that they handle the
extreme ranges a bit better.
Align the C and vector instructions.
Reactivate the unit test asserts when a conversion fails.
Do the channel remapping to the cannonical format when we
deinterleave/interleave instead. Otherwise we would completely skip
the remapping when we have interleaved input.
Fixes#2502, #2490
When we get something else that a drain status as input, bring us back
to the non-drained state.
When we are draining, don't remove the drained flag on the input
io status. This needs to be cleared by the host when the draining is
finished.
Fixes speaker-test
Ensure that our temporary buffers can hold at least quantum_limit
samples. When no output or input is connected, we can generate up
to a quantum_limit of silence, which requires all the buffers to
be scaled correctly.
Fixes a segfault in mpv.
Make a new uint42_t and int24_t type and use that to handle 24 bits
samples. This makes it easier because we can iterate and copy the
structs like other types.
Make dither noise as a value between -0.5 and 0.5 and add this
to the scaled samples.
For this, we first need to do the scaling and then the CLAMP to
the target depth. This optimizes to the same code but allows us
to avoid under and overflows when we add the dither noise.
Add more dithering methods.
Expose a dither.method property on audioconvert. Disable dither when
the target depth > 16.
We need to do dithering and noise when converting f32 to the
target format. This is more natural because we can work in 32 bits
integers instead of floats.
This will also make it possible to actually calculate the error between
source and target values and implement some sort of feedback and
noise shaping later.
Move the noise setting in the dither struct so that it can be
handled separately.
Setup dither separately.
Set used cpu_flags in structures after setup.
The quantize is the amount of bits we want to keep from the original
signal, subtract the amount of bits for noise. Clamp this to 0 (all
noise).
Calculate the scale factor better with powf() and avoid overflows.
Fixes#2479
